• The Journal of the Acoustical Society of Korea
• /
• v.39 no.5
• /
• pp.490-496
• /
• 2020

We present a method to improve the performance of the multi-channel Wiener filter in noisy environment. To build subspace-based multi-channel Wiener filter, in the case of single target source, the target speech component can be effectively estimated in the principal subspace of speech correlation matrix. The speech correlation matrix can be estimated by subtracting noise correlation matrix from signal correlation matrix based on the assumption that the cross-correlation between speech and interfering noise is negligible compared with speech correlation. However, this assumption is not valid in the presence of strong interfering noise and significant error can be induced in the principal subspace accordingly. In this paper, we propose to adjust the principal subspace vector using speech presence probability and the steering vector for the desired speech source. The multi-channel speech presence probability is derived in the principal subspace and applied to adjust the principal subspace vector. Simulation results show that the proposed method improves the performance of multi-channel Wiener filter in noisy environment.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.6
• /
• pp.427-435
• /
• 2016

Array Gain (AG) is a metric to measure the performance of an array of acoustic sensors. AG is affected by the configuration of array, frequency and array element spacing, and the directivity of the ambient noise. In this paper, an algorithm to calculate AG based on the spatial coherence is used, and the results are verified through sea-going experiment. The method using the spatial coherence can be used to consider the arbitrary shape of an array and directionality of ambient noise. In the sea-going experiment, the towed source was used to transmit the Continuous Wave (CW), and was received at the horizontal line array on the seabed. The ambient noise was measured between the source transmission. The experimental AG was calculated from the SNR (Signal to Noise Ratio) of single sensor and an array of sensors. Finally, the predicted AG is shown to agree with the experimental value of AG.

• Geophysics and Geophysical Exploration
• /
• v.9 no.1
• /
• pp.102-107
• /
• 2006

Recently, rapid developments in computer hardware have enabled reverse-time migration to be applied to various production imaging problems. As a wave-equation technique using the two-way wave equation, reverse-time migration can handle not only multi-path arrivals but also steep dips and overturned reflections. However, reverse-time migration causes unwanted artefacts, which arise from the two-way characteristics of the hyperbolic wave equation. Zero-lag cross correlation with diving waves, head waves and back-scattered waves result in spurious artefacts. These strong artefacts have the common feature that the correlating forward and backward wavefields propagate in almost the opposite direction to each other at each correlation point. This is because the ray paths of the forward and backward wavefields are almost identical. In this paper, we present several tactics to avoid artefacts in shot-domain reverse-time migration. Simple muting of a shot gather before migration, or wavefront migration which performs correlation only within a time window following first arriving travel times, are useful in suppressing artefacts. Calculating the wave propagation direction from the Poynting vector gives rise to a new imaging condition, which can eliminate strong artefacts and can produce common image gathers in the reflection angle domain.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.269-272
• /
• 2004

주행 상태에서의 자동차 내부 음 환경은 다양한 소음 및 구조적 요인으로 인하여 음성에 대한 정상적인 전송 특성을 갖기 어렵다. 이는 음원으로부터 음성 입력 장치(Microphone)에 이르기까지의 채널 왜곡에 기인한 문제로써, 실제 주행 환경에서의 음성 인식 성능에 대해서도 심각한 악영향을 초래한다. 본 논문에서는 주행 소음의 크기에 따른 채널별 음성 왜곡 정도에 따른 명료도를 음성 전달 지수인 STI(Speech Transmission Index)를 통하여 분석하고 그 결과를 음성 인식률과 상호 비교하였다. 그리고 수음 패턴에 따른 명료도 척도와 음성 인식 성능과의 상관성을 검토하고, 이를 통해 단일 채널 환경에서 최적의 마이크로폰 위치에 대하여 고찰해 보았다. 실험 결과, 주행 중의 소음 환경에서도 음성의 명료도 척도와 인식률과의 관계는 높은 상관성이 얻어짐을 알 수 있었고, 각 채널 간의 성능 편차 패턴도 주행 환경에 따라 비슷한 양상을 보이는 것으로 나타났다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2011.07a
• /
• pp.254-257
• /
• 2011

본 논문에서는 내용 기반 음악의 감성추출 연구의 동향을 살펴보고, 평가치를 기반으로 청중의 기호를 분석해 본다. 청중의 기호 분석은 가능한 많은 사람이 공감 할 수 있는 데이터를 사용하고자, 최근 이슈가 되고 있는 "나는 가수다" 방송 프로그램의 음원 분석을 통해 이루어졌다. 데이터베이스는 총 7 번의 공연/경연마다 7 곡씩 순위가 매겨지므로 총 49 곡으로 이루어진다. 청중 평가단은 총 500 명의 엄격히 선발된 10 대에서 50 대, 각각 100 명 씩으로 구성되었다. 특징값은 오픈소스인 MIRtoolbox 를 이용해서, 총 376 개의 특징값을 추출하여 평가치와의 상관관계를 구해 청중 평가단의 기호를 분석한다. 실험결과 총 376 개의 특징값 중 10 개의 특징값이 청중평가단의 기호와 상관관계가 있다는 것이 나타났다. 마지막으로 내용 기반 음악의 감성 추출 연구에서 보이는 감성의 주관적 인식성 및 심리학적 설명의 난해함을 줄이고자, 향후 과제로 내용 기반 및 평가치 기반의 시스템을 결합한 감성 기반 음악 추천 시스템을 제안한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2012.07a
• /
• pp.160-163
• /
• 2012

업믹스(Upmix) 기술은 홈시어터와 같은 다채널 스피커 재생 환경에서 콘텐츠의 대부분을 차지하는 스테레오 음원을 다채널 환경에 재생하기 위한 채널 포맷 변환 기술을 말한다. 업믹스를 위한 전처리 단계로서 특정 방향으로 패닝된 주(primary)성분과 잔향 및 배경음과 같은 Ambient 성분을 분리하는 과정이 필요하다. Primary와 Ambient를 분리하기 위한 방법으로 채널 간의 상관도, 적응 필터 및 주성분 분석법(principal component analysis, PCA)이 널리 이용되고 있다. 이에 본 논문에서는 비교적 정확하게 Primary와 Ambient를 분리한다고 알려진 주성분 분석법을 이용하여 신호를 분리해 내고 이 때 주성분 분석법이 가지는 문제점을 해결한 향상된 Primary-Ambient 분리 알고리즘을 제안하였다. 제안된 알고리즘은 분리 성능이 Primary 성분이 패닝된 각도에 영향을 받지 않으며 또한 Primary 성분에 섞인 잔여 Ambient를 제거함으로써 기존의 주성분 분석법 보다 더 정확하게 Primary와 Ambient를 분리 할 수 있고 상관성이 없는 Ambient 특성을 좀 더 정확하게 반영한다.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.18 no.6
• /
• pp.1302-1309
• /
• 2014

In this paper, we propose a method to improve the performance of the direction-of-arrival (DOA) estimation of a speech source using a multiple signal classification (MUSIC)-based algorithm. Basically, the proposed algorithm utilizes a complex coefficient band pass filter to generate the narrow band signals for signal analysis. Also, reverberation component reduction and quadratic function-based response approximation in MUSIC spatial spectrum are utilized to improve the accuracy of DOA estimation. Experimental results show that the proposed method outperforms the well-known generalized cross-correlation (GCC)-based DOA estimation algorithm in the aspect of the estimation error and success rate, respectively.Abstract should be placed here. These instructions give you guidelines for preparing papers for JICCE.

• Journal of Broadcast Engineering
• /
• v.23 no.4
• /
• pp.519-525
• /
• 2018

This paper deals with noise suppression of multi-channel data captured by microphone array using multi-channel Wiener filter. Multi-channel Wiener filter does not rely on information about the direction of the target speech and can be partitioned into an MVDR (Minimum Variance Distortionless Response) spatial filter and a single channel spectral filter. The acoustic transfer function between the single speech source and microphones can be estimated by subspace decomposition of multi-channel Wiener filter. The errors are incurred in the estimation of the acoustic transfer function due to the errors in the estimation of correlation matrices, which in turn results in speech distortion in the MVDR filter. To alleviate the speech distortion in the MVDR filter, diagonal loading is applied. In the experiments, database with seven microphones was used and MFCC distance was measured to demonstrate the effectiveness of the diagonal loading.

• The Journal of the Acoustical Society of Korea
• /
• v.7 no.4
• /
• pp.5-12
• /
• 1988

In order to know the preferred delay time of single reflection in relation in relation to the source signal, and to investigate whether or not there is any display in preference judgment of sound field between subjects of different nationalities, tests of subjective preference for musical sound fields with single reflection were preformed. The result showed that the preferred delay times agreed with the effective duration of auto-correlation function of the source signals, when the amplitude of reflection relative to the direct sound is 0dB. No fundamental disparity in series of judgement of sound field was found even for different series of Judgment with different music motifs. The result of preference test using different passages in single music showed that the fluctuation of the effective duration autocorrelation function over all the passages of the music was small. Thus, the preferred delay time can be determined by the coherence of autocorrelation function of the source signals and the amplitued of reflection.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.19 no.4
• /
• pp.355-362
• /
• 2009

Sound source localization method based on the time delay of arrival(TDOA) is applied to many research fields such as a robot auditory system, teleconferencing and so on. When multi-microphones are utilized to localize the source in 3 dimensional space, the conventional localization methods based on TDOA decide the actual source position using the TDOAs from all microphone arrays and the detection measure, which represents the errors between the actual source position and the estimated ones. Performance of these methods usually depends on the number of microphones because it determines the resolution of an estimated position. In this paper, we proposed the localization method using spatially mapped GCC functions. The proposed method does not use just TDOA for localization such as previous ones but it uses spatially mapped GCC functions which is the cross correlation function mapped by an appropriate mapping function over the spatial coordinate. A number of the spatially mapped GCC functions are summed to a single function over the global coordinate and then the actual source position is determined based on the summed GCC function. Performance of the proposed method for the noise effect and estimation resolution is verified with the real environmental experiment. The mean value of estimation error of the proposed method is much smaller than the one based on the conventional ones and the percentage of correct estimation is improved by 30% when the error bound is ${\pm}20^{\circ}$.